Hi masters, I’m not an expert on this my friends, but I’m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario:
Caller —> GSM Gateway with SIM card A —> Asterisk queue —> extension 1000 GSM gateway with call waiting activated on SIM A Queue with “skip busy agent” disabled and ringall strategy. SIP extension 1000 with call waiting activated, and member of Asterisk queue. a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to Asterisk queue where SIP extension 1000 answers. b) New Caller calls the same SIM card A of GSM gateway (it has call waiting activated on the sim card), call is forwarded to Asterisk queue to the same extension 1000 and a pop-up appears with the second call. c) extension 1000 accepts it so put on hold first call, then try to pickup the new one. The thing is that the SIP re-invite with sendonly attribute can be seen from extension 1000 to Asterisk queue, but this SIP invite is not being forwarded to GSM gateway. So the GSM gateway keeps waiting for it and because it never appears the 1st call is dropped. Maybe you have had this issue in the past. I know that Im not an expert, but I have been researching a lot and trying to vary configurations without clues. The question is: Is it expected for the Asterisk queue to redirect this on-hold message (SIP re-invite with sendonly media attribute) to the GSM gateway so it can manage it call waiting feature on the same SIM card? If we repeat the same scenario without queue intervention (i.e. call goes directly to the extension) the SIP re-invite floods normally between Asterisk and GSM gateway, so GSM gateway can decide what to do with the call. I have no specific queue configuration, seems that queues.conf does not have any parameter to allow this behaviour of re-sending re-invite/on-hold messages. Vendor from GSM gateway side is pointing that “Asterisk js not resending on-hold message”. Thanks and sorry if my ignorance on this,
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
